Summary:ASTERISK-08477: Transfering of calls does not work in 1.4 through chan_agent
Reporter:Peter Zieba (pzieba)Labels:
Date Opened:2007-01-03 22:19:02.000-0600Date Closed:2007-02-16 19:22:20.000-0600
Versions:Frequency of
Description:When trying to transfer a call, only one side of audio is present after tranfer is complete.

Using phones "A" "B" and "C", the situation is illustrated as follows:
"A" calls "B"
"A" & "B" can hear eachother
"B" attempts to tranfer to "C"
"B" & "C" can hear eachother. "A" listens to on-hold music.
"B" Completes transfer
"A" can hear "C". "C" cannot hear "A"

Blind tranfers seem to work properly.

If a call that comes in on a queue (AgentCallBackLogin is being used) is transfered, Asterisk goes into an inconsistent state, while printing messages such as:
[Jan  3 21:27:50] WARNING[14020] chan_zap.c: Ring requested on channel 0/1 already in use on span 1.  Hanging up owner.

No further calls will be taken from the PRI past this point until Asterisk is restarted.

If a call that comes in on a queue is blind-tranfered, things seem to work properly.


CentOS 4.4 (Final)
09:04.0 Communication controller: Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02)
Dell Poweredge 2850 2.8GHz 2GB RAM
Polycom IP430 Phones
Comments:By: Peter Zieba (pzieba) 2007-01-03 22:26:42.000-0600

All phones are on the same network. Believe problem occured in 1.4 (compiled tarball). As far as I can remember, problem was not present in 1.2.10 (although there were problems with call queues in 1.2.10 using agentcallback login, tranfer functionality unreleated to queues worked properly)

By: Peter Zieba (pzieba) 2007-01-03 22:28:06.000-0600

Calls coming in from queue are coming from the TE410P.

Example illustrating phones A, B, and C only includes Sip phones on a dedicated VoIP lan.

By: Peter Zieba (pzieba) 2007-01-03 22:50:27.000-0600

Ignoring the queue aspect of the problem, this seems to be a duplicate of issue 0008696

By: Serge Vecher (serge-v) 2007-01-04 08:20:53.000-0600

alright, since you are using sip devices, let's verify what the problem is by producing the log as outlined in note 0057086 in 8696

By: Peter Zieba (pzieba) 2007-01-04 16:08:29.000-0600

I can try testing some of this later tonight while there are no call coming in so I can provide you guys with some logs, but I cannot simulate this setup as I do not have an extra PRI to play with.

I will be rolling back to Asterisk 1.2.14 at midnight.

By: Joshua C. Colp (jcolp) 2007-01-25 21:30:44.000-0600

Any update on this? Logs/whether it is solved/etc?

By: Joshua C. Colp (jcolp) 2007-02-16 19:22:20.000-0600

Fixed in 1.2 as of revision 55073, 1.4 as of revision 55086, and trunk as of revision 55087. Peace!